European ETS Telecommunication July 1999 Standard

Transcription

1 European ETS Telecommunication July 1999 Standard Sixth Edition Source: SMG Reference: RE/SMG PR4 ICS: Key words: Digital cellular telecommunications system, Global System for Mobile communications (GSM) GLOBAL SYSTEM FOR MOBILE COMMUNICATIONS Digital cellular telecommunications system (Phase 2); Transmission planning aspects of the speech service in the GSM Public Land Mobile Network () system (GSM version 4.6.1) R ETSI European Telecommunications Standards Institute ETSI Secretariat Postal address: F Sophia Antipolis CEDEX - FRANCE Office address: 650 Route des Lucioles - Sophia Antipolis - Valbonne - FRANCE Internet: secretariat@etsi.fr - Tel.: Fax: Copyright Notification: No part may be reproduced except as authorized by written permission. The copyright and the foregoing restriction extend to reproduction in all media. European Telecommunications Standards Institute All rights reserved.

2 Page 2 Whilst every care has been taken in the preparation and publication of this document, errors in content, typographical or otherwise, may occur. If you have comments concerning its accuracy, please write to "ETSI Standards Making Support Dept." at the address shown on the title page.

3 Page 3 Contents Intellectual Property Rights... 7 Foreword Scope Normative references Definitions and abbreviations Introduction Network configurations General Model of the Interfaces Configurations of Connections General Configurations of Connections Reference configurations to illustrate delay and echo control issues wire circuits in the Transmission performance Overall Loss/Loundness ratings Connections with handset MSs Connections with handsfree MSs using loudspeakers Connections with headset MSs Stability Loss Delay General Sources of delay Elements of the that cause delay Elements of the PSTN that cause delay Effects of delay Allocation of delay to the Allocation of delay to the when using a full rate system Allocation of delay to the when using a half rate system Delay of various network configurations National and international connections with no echo control in the PSTN (reference configurations A) National and international connections with echo control in the PSTN (reference configurations B) Connections where re-routeing leads to a significant increase in transmission path length (reference configurations C) Delay related requirements on the MS Full rate MS Half rate MS Echo General Electrical echo control in the (Reference configurations A) Acoustic echo control in the Acoustic echo control in a handsfree MS Acoustic echo control in a handset MS Acoustic echo control in a headset MS Interaction between tandem echo control devices (reference configurations B & C) Clipping... 21

4 Page General Properties of voice switches in the Problems of tandem voice switching Idle channel noise Sending Receiving Noise contrast General Elements of a which can cause noise contrast impairment Reduction of noise contrast Reduction of noise contrast by limiting the noise received by the microphone Headset MS Handset MS Handsfree MS Reduction of noise contrast by insertion of comfort noise Consequence of the introduction of high comfort noise levels on other voice-operated devices Sensitivity/frequency characteristics Headset and Handset MSs Sending Receiving Handsfree MS Distortion Sending Receiving Sidetone Sidetone loss Sidetone distortion Out-of-band signals Discrimination against out-of-band input signals Spurious out-of-band signals Requirements for information tones Crosstalk Near and far end crosstalk Go/return crosstalk Annex A (informative): Considerations on the Acoustic Interface of the Mobile Station A.1 Handsfree MS A.2 Handset MS A.3 Headset MS A.4 Inter-reaction with DTX Annex B (normative): Transmission requirements testing B.1 Loudness ratings B.1.1 Sending Loudness Rating (SLR) B.1.2 Receiving Loudness Rating (RLR) B.2 Idle Channel Noise B.2.1 Sending B.2.2 Receiving B.3 Sensitivity/frequency Characteristics B.3.1 Sending B.3.2 Receiving B.4 Distortion... 42

5 Page 5 B.4.1 Sending B.4.2 Receiving B.5 Variation of gain with input level B.5.1 Sending B.5.2 Receiving B.6 Sidetone B.6.1 Talker sidetone (STMR) B.6.2 Listener sidetone (LSTR) B.7 Sidetone distortion B.8 Out-of-band signals B.8.1 Discrimination against out-of-band input signal B.8.2 Spurious out-of-band signals B.9 Acoustic echo loss Annex C (normative): MS delay requirement definition C.1 Full rate MS delay requirement definition C.2 Half rate MS delay requirement definition Annex D (informative): Change Request History History... 50

6 Page 6 Blank page

7 Page 7 Intellectual Property Rights IPRs essential or potentially essential to the present document may have been declared to ETSI. The information pertaining to these essential IPRs, if any, is publicly available for ETSI members and nonmembers, and can be found in SR : "Intellectual Property Rights (IPRs); Essential, or potentially Essential, IPRs notified to ETSI in respect of ETSI standards", which is available free of charge from the ETSI Secretariat. Latest updates are available on the ETSI Web server ( Pursuant to the ETSI IPR Policy, no investigation, including IPR searches, has been carried out by ETSI. No guarantee can be given as to the existence of other IPRs not referenced in SR (or the updates on the ETSI Web server) which are, or may be, or may become, essential to the present document. Foreword This European Telecommunication Standard (ETS) has been produced by the Special Mobile Group (SMG) Technical Committee of the European Telecommunications Standards Institute (ETSI). This ETS describes the transmission planning aspects pertaining to the speech service within the digital cellular telecommunications system (Phase 2). The contents of this ETS is subject to continuing work within SMG and may change following formal SMG approval. Should SMG modify the contents of this ETS, it will be resubmitted for OAP by ETSI with an identifying change of release date and an increase in version number as follows: Version 4.x.y where: 4 indicates GSM Phase 2 x y the second digit is incremented for changes of substance, i.e. technical enhancements, corrections, updates, etc. the third digit is incremented when editorial only changes have been incorporated in the specification. Transposition dates Date of adoption of this ETS: 23 July 1999 Date of latest announcement of this ETS (doa): 31 October 1999 Date of latest publication of new National Standard or endorsement of this ETS (dop/e): 30 April 2000 Date of withdrawal of any conflicting National Standard (dow): 30 April 2000

8 Page 8 Blank page

9 Page 9 1 Scope This European Telecommunication Standard (ETS) is concerned with the transmission planning aspects pertaining to the speech service in the pan-european system. Due to technical and economic factors, there cannot be full compliance with the general characteristics of international telephone connections and circuits recommended by CCITT. This ETS gives guidance as to the precautions, measures and minimum requirements needed for successful interworking of the with the national and international PSTN. The Recommendation identifies a number of routeing and network configurations. The objective is to reach a quality as close as possible to CCITT standards in order to safeguard the performance seen by PSTN customers. 1.1 Normative references This ETS incorporates by dated and undated reference, provisions from other publications. These normative references are cited at the appropriate places in the text and the publications are listed hereafter. For dated references, subsequent amendments to or revisions of any of these publications apply to this ETS only when incorporated in it by amendment or revision. For undated references, the latest edition of the publication referred to applies. Unless otherwise stated, all references to CCITT Recommendations are from the Blue Book (1989). [1] GSM (ETR 100): "Digital cellular telecommunication system (Phase 2); Abbreviations and acronyms". [2] GSM (ETS ): "Digital cellular telecommunication system (Phase 2); Signalling requirements relating to routeing of calls to mobile subscribers". [3] GSM (ETS ): "Digital cellular telecommunication system (Phase 2); Full rate speech processing functions". [4] GSM (ETS ): "Digital cellular telecommunication system (Phase 2); Full rate speech transcoding". [5] GSM (ETS ): "Digital cellular telecommunication system (Phase 2); Substitution and muting of lost frames for full rate speech channels". [6] GSM (ETS ): "Digital cellular telecommunication system (Phase 2); Comfort noise aspect for full rate speech traffic channels". [7] GSM (ETS ): "Digital cellular telecommunication system (Phase 2); Discontinuous Transmission (DTX) for full rate speech traffic channel". [8] GSM (ETS ): "Digital cellular telecommunication system (Phase 2); Voice Activity Detection (VAD)". [9] GSM (ETS ): "Digital cellular telecommunication system (Phase 2); Half rate speech processing functions". [10] GSM (ETS ): "Digital cellular telecommunication system (Phase 2); Half rate speech transcoding". [11] GSM (ETS ): "Digital cellular telecommunication system (Phase 2); Substitution and muting of lost frames for half rate speech traffic channels". [12] GSM (ETS ): "Digital cellular telecommunication system (Phase 2); Comfort noise aspects for half rate speech traffic channels".

10 Page 10 [13] GSM (ETS ): "Digital cellular telecommunication system (Phase 2); Discontinuous Transmission (DTX) for half rate speech traffic channels". [14] GSM (ETS ): "Digital cellular telecommunication system (Phase 2); Voice Activity Detection (VAD) for half rate speech traffic channels". [15] ETS (1990): "Integrated Services Digital Network (ISDN); 3,1 khz telephony teleservice; Attachment requirements for handset terminals". [16] CCITT Recommendation G.103: "Hypothetical reference connections". [17] CCITT recommendation G.111: "Loudness ratings (LRs) in an international connections". [18] CCITT Recommendation G.113: "Transmission impairments". [19] CCITT Recommendation G.114: "Mean one-way propagation time". [20] CCITT Recommendation G.121: "Loudness ratings (LRs) of national systems". [21] CCITT Recommendation G.122: "Influence of national systems on stability, talker echo, and listener echo in international connections". [22] CCITT Recommendation G.131: "Stability and echo". [23] CCITT Recommendation G.165: "Echo cancellers". [24] CCITT Recommendation G.223: "Assumptions for the calculation of noise on hypothetical reference circuits for telephony". [25] CCITT Recommendation G.703: "Physical/electrical characteristics of hierarchical digital interfaces". [26] CCITT Recommendation G.711: "Pulse code modulation (PCM) of voice frequencies". [27] CCITT Recommendations G.712: "Transmission performance characteristics of pulse code modulation". [28] CCITT Recommendations G.714: "Separate performance characteristics for the send and receive sides of PCM channels applicable to 4-wire voice frequency interfaces". [29] CCITT Recommendations M.1020: "Characteristics of special quality". [30] CCITT Recommendations M.1025: "Characteristics of special quality international leased circuits with basic bandwidth conditioning". [31] CCITT Recommendations M.1030: "Characteristics of ordinary quality international leased circuits forming part of private switched telephone networks". [32] CCITT Recommendations M.1040: "Characteristics of ordinary quality international leased circuits". [33] CCITT Recommendation O.132 (1988), "Specification for a quantizing distortion measuring apparatus using a sinusoidal test signal." [34] CCITT Recommendation P.11: "Effect of transmission impairments".

11 Page 11 [35] CCITT Recommendation P.34: "Transmission characteristics of hands-free telephones". [36] CCITT Recommendation P.38: "Transmission characteristics of operator telephone systems (OTS)". [37] CCITT Recommendation P.50: "Artificial voices". [38] CCITT Recommendation P.51 (1988), "Artificial mouths and artificial ears." [39] CCITT Recommendation P.64 (1988), "Determination of sensitivity/frequency characteristics of local telephone systems to permit calculation of their loudness ratings." [40] CCITT Recommendation P.76 (1988), "Determination of loudness ratings; fundamental principles." [41] CCITT Recommendation P.79 (1988), "Calculation of loudness ratings." [42] CCITT Recommendation Q.35: "Technical characteristics of tones for the telephone service". [43] CCITT Recommendation Q.551: "Transmission characteristics of digital exchanges". [44] CCITT Blue Book (1988), Volume V, Supplement 13, "Noise spectra." [45] ISO , "Preferred numbers - series of preferred numbers." [46] ITU-T Recommendation P.57 (1996), "Artificial ears." 1.2 Definitions and abbreviations In addition to those below the definitions and abbreviations used in this specification are listed in GSM ADC ADPCM AEC BSC' BTS' DAC DMR DSI EEC EL ERP FDM LSTR MRP OLR PCM POI RLR SLR STMR UPCMI Analogue to Digital Converter Adaptive Differential Pulse Code Modulation Acoustic Echo Control Base Station Controller (excluding transmission sytems Base Tranceiver Station (excluding transmission systems) Digital to Analogue Converter Digital Mobile Radio Digital Speech Interpolation Electric Echo Control Echo Loss Ear Reference Point Frequency Division Multiplex Listener Sidetone Rating Mouth Reference Point Overall Loudness Rating Pulse Code Modulation Point of Interconnection (with PSTN) Receiver Loudness Rating Send Loudness Rating Sidetone Masking Rating 13-bit Uniform PCM Interface

12 Page Introduction Since the transmission quality and the conversational quality of the will in general be lower than the quality of the PSTN connection due to coding distortion, delay, etc, only some transmission aspects can be brought in line with CCITT Recommendations. It is therefore necessary to improve the overall quality as much as possible by implementing proper routeing and network configurations. It should be recognised that the transmission plan for the pan-european cannot lead to major changes in the PSTN. However, it is important to use the improvements in the evolving PSTN (e.g. digitalization, introduction of echo cancellers) in an effective way. The transmission requirements are in the first place based on international connections. When the quality is sufficient for international connections, it can be assumed that the national connections will have the same or better quality. In order to obtain a sufficient quality in the connection, it is preferable to have digital connectivity between the Base Station System (BSS) and the international exchange. The requirements are based on this assumption. When this situation cannot be provided, a lower quality must temporarily be accepted. This Recommendation consists of two parts: one will deal with network configurations, the other with transmission performance. The part about network configurations gives information about the reference connections, on which the transmission plan is based. Furthermore, some guidelines are presented for improvement of the transmission quality in the evolving (digital) PSTN. The part about transmission performance gives mainly characteristics of the transmission between MS acoustic interface (MRP/ERP) and the interface between the and the PSTN (POI). For transmission aspects where it is impossible to give overall characteristics, it is in some cases necessary to make recommendations for individual parts of the equipment. Unless otherwise stated, all references to CCITT Recommendations are from the Blue Book (1989). Annex A considers the effects of the type of acoustic interfaces of the MS. 2 Network configurations 2.1 General The basic configuration for the interworking with the PSTN is shown in figure Model of the A more detailed model of the used for the consideration of transmission planning issues for speech is shown in figure 2. This model represents the main functions required and does not necessarily imply any particular physical realisation. Routeing of calls is given in Recommendation GSM Any acoustic echo control is not specifically shown as it will be provided by analogue processing of digital processing or a combination of both techniques.

13 Page Interfaces The main interfaces identified within the GSM Recommendations are shown in figure 1. For the purposes of this Recommendation, the Air Interface and the Point of Interconnect (POI) are identified along with two other interfaces, Interface Z and a 13-bit Uniform PCM Interface (UPCMI). These interfaces are needed to define the transmission characteristics and the overall system requirements. The Air Interface is specified by GSM 05 series Recommendations and is required to achieve MS transportability. Analogue measurements can be made at this point by using the appropriate radio terminal equipment and speech transcoder. The losses and gains introduced by the test speech transcoder will need to be specified. The POI with the PSTN will generally be at the 2048 kbits/s level at an interface, in accordance with CCITT Recommendation G.703. At the point, which is considered to have a relative level of O dbr, the analogue signals will be represented by 8-bit A-law, according to CCITT Recommendation G.711. Analogue measurements may be made at this point using a standard send and receive side, as defined in CCITT Recommendations G.714 and G.712. Interface Z might be used in the case of direct MSC to MSC connections. Interface Z is of the same nature as the POI. The UPCMI is introduced for design purposes in order to separate the speech transcoder impairments from the basic audio impairments of the MS. 2.4 Configurations of Connections General Configurations of Connections Figure 3 shows a variety of configurations of connections. There are a number of PSTN features which should be avoided from such connections. These include: - echo control devices in the international network. If present, and not disabled, these devices will be in tandem with echo cancellers and may introduce degradation; - satellite routeings. The delay inherent in the connections when added to the delay, may result in conversational difficulties. Double satellite links are likely to cause severe difficulties and special precautions should be taken to avoid this situation under call forwarding arrangements; - digital speech interpolation systems (DSI). There is likely to be an adverse interaction between DSI and DTX; - ADPCM. The distortion introduced by ADPCM on routes where PSTN echo control is not provided is likely to reduce the echo cancellation provided by the electric echo canceller; - significant differences in clock rates on non-synchronised digital network components. The resulting phase roll and slips are likely to degrade the performance of the echo canceller; - those analogue FDM routeings which exhibit phase roll. Any phase roll due to the absence of synchronisation between the carrier frequencies on the two directions of transmission is likely to degrade the performance of the echo canceller; - tandem connections of sources of quantisation distortion. The speech transcoder is estimated to be equivalent to 7 QDUs between uniform PCM interfaces (see CCITT Recommendation G.113).

14 Page 14 It is recognised that on some connections it may not be feasible to avoid these features, but in many cases, especially if taken into account at the planning stage, this should be possible Reference configurations to illustrate delay and echo control issues Three basic reference configuration types shown in Figures 4 to 6 are defined to illustrate delay and echo control issues. Intermediate echo control devices as shown in the figures are disabled by appropriate signalling between the MSC and ISC or MSC and MSC. Reference configurations A (see figure 4) represent national or international connections where there is no echo control device in the PSTN. These reference configurations include re-routeing configurations where the overall delay of the transmission path has not been extended. Reference configurations B (see figure 5) represent national or international connections where echo control is provided in the PSTN. These reference configurations include re-routeing configurations where the overall delay of the transmission path has not been extended. Reference configurations C (see figure 6) represent national or international connections where rerouteing has lead to an increase in the overall delay of the transmission path beyond recommended limits wire circuits in the As shown in figure 2, the will usually contain transmission systems. Where present, they should provide 4-wire circuits. In the case of digital circuits which do not include any speech processing devices, the overall system requirements of the will not be affected by the presence of the link. In the case of analogue links, the transmission characteristics (e.g. attenuation, attenuation distortion, noise) will affect the overall system requirements of the. CCITT Recommendations M.1020, M.1025, M.1030 and M.1040 describe several transmission characteristics for leased circuits. In cases where the analogue link introduces loss, provision will have to be made at the interface to restore the loss. 3 Transmission performance The overall transmission performance of connections in alternate conversation mode can be considered as a summation of the effects of: - the audio part between the MRP/ERP and the UPCMI interface; - the speech transcoder part including the effects of radio transmission, and speech processing between the UPCMI and the POI; - the overall characteristics of the connection between POI and the other user. There is not only a linear addition of these effects but there is also an influence from different parts of the connection on the performance of the speech transcoder and other speech processing devices. Where possible, the transmission performance is specified between the MRP/ERP and the POI. Where this is not possible, the transmission aspects of the audio part mentioned above have been specified. The transmission aspects of the speech transcoder are specified in GSM 06 series Recommendations. In the following subclauses, requirements are specified for the UPCMI, the Air Interface or the POI as appropriate.

15 Page 15 The following subclauses are applicable to handset MSs. In some places, reference is made to headset and handsfree MSs, but further study is needed to fully extend this Recommendation to these types of acoustic interface (see Annex A). The transmission requirements of the MS have been derived from the requirements of digital telephones stated in ETS (December 1990). MSs will have to work in a variety of environments ranging from quiet office locations to very noisy environments as found in moving cars. In noisy conditions, different values for SLR, STMR and low frequency response may be required. These different values may be achieved by introducing some switch-over function (manual or automatic). This point needs further study. The overall transmission performance in full duplex conversation mode will also greatly depend on the performance of the echo control devices which may be included in the connection. 3.1 Overall Loss/Loundness ratings The overall international connection involving s and the PSTN should meet the overall loundness rating (OLR) limits in CCITT Recommendation G.111. The national parts of the connection should therefore meet the send and receive loudness rating (SLR, RLR) limits in CCITT Recommendation G.121. For the case where digital routeings are used to connect the to the international chain of circuits, the SLR and RLR of the national extension will be largely determined by the SLR and RLR of the. The limits given below are consistent with the national extension limits and long term objectives in CCITT Recommendation G.121. The SLR and RLR values for the apply up to the POI. However, the main determining factors are the characteristics of the MS, including the analogue to digital conversion (ADC) and digital to analogue conversion (DAC). Hence, in practice, it will be convenient to specify loudness ratings to the Air Interface. For the normal case, where the introduces no additional loss between the Air Interface and the POI, the loudness ratings to the PSTN boundary (POI) will be the same as the loudness ratings measured at the Air Interface. However, in some cases loss adjustment may be needed for interworking situations in individual countries. These values are directly applicable to the case of an MS operating in a conventional non-mobile noise environment. Studies have shown that under the noise environment, speech levels are likely to be higher. Hence, in order to avoid clipping in the speech transcoder, the value of SLR may need to be increased. NOTE: Measurement of SLR and RLR using sinusoidal test frequencies may not be sufficiently accurate because of the adaptive characteristics of the full-rate speech transcoder. A possible method is to use the artificial voice described in CCITT Recommendation P.50 to measure send and receive sensitivities. A method used by one administration uses the artificial voice to measure the loudness rating according to the Zwicker algorithm Connections with handset MSs The nominal values of SLR/RLR to the POI shall be: SLR = 8 +/- 3 db; RLR = 2 +/- 3 db. Where a user-controlled receiving volume control is provided, the RLR shall meet the selected nominal value for at least one setting of the control. When the control is set to maximum, the RLR shall not be less than (louder than) - 13 db.

16 Page 16 With the volume control set to the minimum position the RLR shall not be greater than (quieter than) 18 db. Compliance shall be checked by the tests described in Annex B, subclauses B.1.1 and B.1.2. NOTE: The mechanical design of some MSs may make it impossible to seal the earpiece to the knife edge of the CCITT artificial ear. Minimal additional methods may be used to provide the seal provided that they do not affect the mounting position of the MS with respect to the Mouth Reference Point and the Ear Reference Point Connections with handsfree MSs using loudspeakers The SLR and RLR should be measured and computed using the methods given in CCITT Recommendation P.34 section 6 with an artificial voice satisfying CCITT Recommendation P.50. The values of SLR/RLR to/from the POI should be: SLR = 13 +/- 3 db (note 1); RLR = 2 +/- 3 db with the volume control set to the mid position. A receive volume control should be provided with a range of between +/- 7.5 db and +/- 15 db. The use of values towards the most sensitive end of the range may result in problems with amplifying crosstalk from other channels (note 2). NOTE 1: NOTE 2: NOTE 3: This values takes into account the CCITT Recommendation P.34; the SLR of a handsfree telephone should be 5 db higher than the corresponding value for a handset instrument. The tolerance of +/- 3 db is provisional. This procedure assumes no automatic gain control in the mobile terminal. The use of such techniques is not recommended for mobile applications. Further work is required to develop a practical test method using CCITT Recommendation P Connections with headset MSs The SLR and RLR should be measured and computed using methods given in CCITT Recommendation P.38. This Recommendation currently gives a measuring technique for supra-aural earphone and inserttype receivers. Study is continuing on other types of earpieces in CCITT SGXII. The nominal values of SLR/RLR to/from the POI should be: SLR = 8 +/- 3 db; RLR = 2 +/- 3 db with any volume control set to mid position. Where a user-controlled receiving volume control is provided, the RLR shall meet the selected nominal value for at least one setting of the control. When the control is set to maximum, the RLR shall not be less than (louder than) - 13 db. With the volume control set to the minimum position the RLR shall not be greater than (quieter than) 18 db. 3.2 Stability Loss The stability loss presented to the PSTN by the at the POI should meet the principles of the requirements in Sections 2 and 3 of CCITT Recommendation G.122. These requirements will be met if the attenuation between the digital input and digital output at the POI is at least 6 db at all frequencies in the range 200 Hz to 4 khz under the worst-case acoustic conditions at the MS (any acoustic echo control should be enabled).

17 Page 17 For the normal case of digital connection between the Air Interface and the POI, the stability requirement can be applied at the Air Interface. The worst-case acoustic conditions will be as follows (with any volume control set to maximum): Handset MS: uncapped. Handsfree MS: study). Headset MS: NOTE: the handset lying on, and the transducers facing, a hard surface with the earpiece a representative worst-case position of microphone and loudspeaker (for further for further study. The test procedure will need to take into account the switching effects of echo control and DTX. 3.3 Delay General A significant propagation time between the two ends of a connection causes difficulties in conversation over the connection. This arises from two causes. Firstly, the signal is reflected back from the distant end causing an echo to the talker (this is considered in subclause 3.4). Secondly, even if ideal echo control were achieved, the delay between a user talking and receiving a reply from the user at the distant end of the connection could cause conversational difficulty. s will be connected to the PSTN at a point where present planning rules allow for a delay of less than 12 ms (see CCITT Recommendation G.114 subclause 2.2a). The delay within the will greatly exceed this. If unacceptable circuit delays are not to be experienced by users, action will have to be taken when planning routes or during call set-up Sources of delay Elements of the that cause delay The delay of the is made up of the following elements: - speech transcoding delay; - radio channel coding delay; - network delay (i.e. fixed elements such as multiplexing, propagation, switching, echo control) Elements of the PSTN that cause delay CCITT recommendation G.114 identifies various elements present in some PSTN connections which cause delay. These include: - coaxial, radio and optical fibre terrestrial transmission systems; - geostationary satellites; - digital speech interpolators; - digital exchanges (see also CCITT recommendation Q.551); - echo cancellers.

18 Page Effects of delay Some recent studies have suggested that under ideal conditions, i.e.: - effective control of all echoes without clipping by the use of good echo cancellers; - low background noise leading to an absence of perceptible noise contrast; - low distortion of transmitted signals; - ideal loudness ratings; users can tolerate a circuit delay well in excess of 400 ms (currently the maximum delay recommended in CCITT Recommendation G.114). Other studies indicate that the difficulty caused by circuit delay increases when impairments, such as imperfect echo control caused by echo suppressers, clipping and noise contrast, are present. However, the mobile environment is very harsh, with high background noise levels and distortion from the speech transcoder. In particular, the use of acoustic echo suppression could give rise to severe speech clipping and noise contrast. Also the operation of the voice switching used with DTX will give impairments similar to those caused by echo suppression. All subjective tests performed with echo suppressors indicate that, because of the increased effect of clipping with increased delay, the difficulty experienced by users increases rapidly with delay. According to curve 2 of figure A.1 of CCITT Recommendation G.114, the percentage of users experiencing difficulties with echo suppressors reaches 20% with a delay of 150 ms rising to 40% with a delay of 300 ms. CCITT Recommendation G.114 Annex A details the test conditions under which this curve was derived and it concludes that connections with more than 300 ms can only be used by very disciplined users who are aware of the problems involved in such a connection. However, recent work has indicated that delays of up to 500 ms can be used satisfactorily, provided that effective echo cancellation is incorporated in the link Allocation of delay to the Allocation of delay to the when using a full rate system Taking account of Recommendations on the separate factors described in subclause , the maximum both-way in the between the MRP/ERP and the Point of Interconnection (see figure 1) will be 180 ms. In the case that the transcoder is positioned outside the BTS, the maximum distance between the POI and the furthest border of the cell controlled by the BTS is limited by a one-way propagation delay of 1.5 ms (approximately 300 km). If the transcoder is positioned at the BTS, the limit is 6.5 ms (approximately 1300 km). These limits may be subject to increase resulting from savings made in the overall network Allocation of delay to the when using a half rate system If it is assumed that the speech quality associated with the half rate system is the same as the full rate system (considering both the speech transcoder and the radio sub-system), then in order to achieve the same overall transmission quality, the maximum delay within the should be maintained at 180 ms Delay of various network configurations National and international connections with no echo control in the PSTN (reference configurations A) Reference configurations A (see figure 4) contain no echo control in the PSTN because present planning rules require the use of echo control devices only when the PSTN delay between two fixed PSTN users exceeds 25 ms. This leads to a maximum PSTN delay of 22 ms from the point of interconnection to the (see subclause 3.4.2).

19 Page National and international connections with echo control in the PSTN (reference configurations B) Reference configurations B (see figure 5) contain echo control in the PSTN because present planning rules require their use when the PSTN delay between PSTN users exceeds 25 ms. However, action may have to be taken by administrations when planning routes or at call set-up to limit the maximum delay. Subclause describes how the impairments from the harsh mobile environment when coupled with delay can give rise to difficulty. If very good cancellation of both electrical and acoustic echo can be achieved and there are no sources of speech clipping or noise contrast either in the or the PSTN part of the connection, the circuit delay should be kept below 400 ms. This means that every attempt should be made to avoid mobile to mobile calls via satellite (expected delay > 440 ms). If acoustic echo suppression is used or DTX is enabled, or there is any other source of clipping or noise contrast present in the PSTN, the additional distortion introduced makes it desirable to avoid any satellite routeing whenever possible in order to keep the delay below 300 ms Connections where re-routeing leads to a significant increase in transmission path length (reference configurations C) A number of possible combinations of re-routeing are described by reference configurations C (see figure 6), all of which increase the path length and hence the delay and some of which increase the number of impairments in the network. These routeings are likely to cause severe degradation to the quality of the connection and may result in significant difficulty, particularly when the connection contains one or maybe more satellite links. These connections should be avoided in network planning and, if this is not possible, then the facilities of Signalling System No. 7 should be used to control the routeing of the call at call set-up to minimise the effects Delay related requirements on the MS Full rate MS In accordance with the outline of transmission delay in various GSM system elements contained in GSM the trip delay in the MS shall not exceed ms as defined in Annex C Half rate MS The round trip delay in the MS shall not exceed ms as defined in Annex C. 3.4 Echo General There are two main sources of echo: - acoustic echo caused by the acoustic path between receive and transmit transducers; - electrical echo caused by coupling between the transmit and receive directions of transmission. The primary source of this form of echo is a two-to-four wire converter. Electrical echo can be eliminated by the use of end-to-end four-wire transmission. Acoustic echo will be generated in all telephone instruments with the exception of carefully designed headsets. In general, electrical echo is characterised by a short reverberation time and low dispersion while acoustic echo is likely to have a longer reverberation time and greater dispersion. The case of the acoustic echo may be further complicated by the time variant nature of acoustic echo which may be more severe in the mobile environment.

20 Page 20 Curves showing the tolerance to echo, taking account of the relationship between the delay and the level of the echo, are given in CCITT Recommendation G.131 figure 2/G.131. In practice, it has been found that for any connection with a delay of greater than 25 ms, some form of echo control will be required to reduce the level of the echo (CCITT Recommendation G.131 Rule M). With the expected maximum one-way delay in the of 90 ms, acoustic echo control will be required in the MS to reduce the echo returned to the distant end and electrical echo control will be required at the POI to reduce the echo returned to the user from the PSTN. The design of these echo control devices should be such as to provide operation in full duplex mode (as opposed to alternate mode). The echo loss (EL) presented by the at the POI should be at least 46 db during single talk. This value takes into account the fact that a MS is likely to be used in a wide range of noise environments. This requirement should be met for both handset and handportable MSs. The requirement for handsfree MSs is for further study. The test method is defined in Annex B, clause B Electrical echo control in the (Reference configurations A) The electrical echo control device at the interface with the PSTN should meet the requirements given in CCITT Recommendation G.165, but with an end delay of 60 ms. This refers to t d in subclause 3.2 of CCITT Recommendation G.165. The 60 ms is calculated as follows. CCITT Recommendation G.131 states that the maximum length of connection which need not have echo control has a mean one-way propagation time of 25 ms. However, this figure is the sum of the delays of the international connection and the maximum national delays at each end of the connection. Since the interconnection of the to the PSTN is unlikely to be at a point where the PSTN delay is > 22 ms, and the dispersion may be up to 8 ms, the maximum expected end delay which the echo canceller in the MSC should expect is: (22 + 8) x 2 = 60 ms (see figure 7). Certain countries on the geographical limits of a continent may need to increase this limit as there may be a proportion of connections which do not comply with CCITT Recommendation G.131 having a mean one-way delay of greater than 25 ms and yet are not provided with echo control Acoustic echo control in the Acoustic echo control provided in the MS should provide an EL of 46 db at the POI (see subclause 3.4.1) over the likely range of acoustic end delays. If acoustic echo control is provided by voice switching, comfort noise should be injected. This comfort noise shall operate in the same way to that used in Discontinuous Transmission system (DTX). Effectively, the acoustic echo loss is provided by MS as the GSM network is zero loss from the air interface to the POI and hence the 46dB requirement shall be applied to the MS Acoustic echo control in a handsfree MS The telephone transmission parameters for handsfree MS are for further study. However, the basic requirement of 46 db echo loss should apply to handsfree terminals, also. If acoustic echo control is provided using some form of echo cancellation technique, the cancellation algorithm should be designed to cope with the expected reverberation and dispersion. In the case of the handsfree MS, this reverberation and dispersion may be time variant. The expected values of dispersion are under study Acoustic echo control in a handset MS The echo loss requirement for the handset MS shall be 46 db. Careful acoustic design of the handset body and selection of the mouth and ear piece transducers may facilitate the required acoustic echo loss without the need for active echo control techniques. However, should echo cancellation be employed the echo canceller should be capable of dealing with the variations in handset positions when in normal use. The implications of this are under study Acoustic echo control in a headset MS The echo loss requirement for a headset shall be 46 db. Due to the obstacle effect of the head in this type of terminal, careful design might mean that no active echo control is necessary.

21 Page Interaction between tandem echo control devices (reference configurations B & C) On long international routes or routes containing a satellite path, network echo control devices will be present in accordance with CCITT Recommendation G.131 Rule M. These devices will be echo suppressors or echo cancellers generally with centre clippers. The tandem connection of such devices can lead to increased clipping and, if echo suppressors are used, additional loss. It is recommended that signalling or routeing means be used to avoid the tandem connections of echo control devices whenever possible (see figure 7). 3.5 Clipping General The loss of the start or the end of a speech burst is known as clipping, the main cause of which is voice switching controlled by voice activity detection. Voice switching occurs in devices within the network or within terminal devices. The following devices employ voice switching: - echo suppressors. These are generally located at an ISC at either end of a long international connection or connections using satellites. - echo cancellers with centre clippers. These are located as for the echo suppressors above. In addition, it is recommended that they be used in the MSC at the interface with the PSTN. Clipping in these devices arises from the action of the centre clipper only. - digital speech interpolators (DSI). These devices are used in circuit multiplication equipments which are often employed on international connections. - discontinuous transmission (DTX) devices. These are located in the. - loudspeaking telephones. These are used in the PSTN and in the. It should be noted that regulations in certain countries prohibit the use of handheld MSs by drivers of moving vehicules Properties of voice switches in the Recommendation GSM specifies the requirements for the voice activity detector used for DTX and the total clipping allowed in the MS. Any voice switching used for acoustic echo control should not exceed these limits. Information on recommended characteristics of handsfree telephones is given in section 5 of CCITT Recommendation P Problems of tandem voice switching The effect of tandem voice switches which are not under one common control will be an increase in clipping. Moreover, under conditions of high or rapidly changing ambient noise, false detection of speech is likely to occur in the voice activity detectors in DSI equipment or network echo control devices. These devices are generally designed for constant and low levels of noise. In order to minimise clipping, the following action should be taken: - intermediate tandem voice switching devices in the network should be either disabled by signalling means or avoided by routeing means; - the voice switching for the MS for acoustic control and for DTX should be under one common control. However, it should be noted that, in many cases, it will not be possible to exclude DSI equipment or loudspeaking telephones from the connection.

22 Page Idle channel noise Sending The maximum noise level produced by the apparatus at the UPCMI under silent conditions in the sending direction shall not exceed - 64 dbm0p. NOTE 1: NOTE 2 This level includes the eventual noise contribution of an acoustic echo canceller under the condition that no signal is received. This figure applies to the wideband noise signal. It is recommended that the level of single frequency disturbances should be 10 db lower (CCITT Recommendation P.11). Compliance shall be checked by the test described in Annex B, subclause B Receiving The maximum (acoustic) noise level at the handset MS when no signal (O-level) is received from the speech transcoder shall be as follows: If no user-controlled receiving volume control is provided, or, if it is provided, at the setting of the user-controlled receiving volume control at which the RLR is equal to the nominal value, the noise measured in the artificial ear contributed by the receiving equipment alone shall not exceed - 57 dbpa(a) when driven by a PCM signal corresponding to the decoder output value number 1. Where a volume control is provided, the measured noise shall also not exceed - 54 dbpa(a) at the maximum setting of the volume control. NOTE: In a connection with the PSTN, noise conditions as described in CCITT Recommendation G.103 can be expected at the input (POI) of the. The characteristics of this noise may be influenced by the speech transcoding process (for further study). Compliance shall be checked by the test described in Annex B, subclause B Noise contrast General On any call there is likely to be continuous background noise which is present regardless of whether the users are talking or not. There may also be one or more voice-operated devices; these effectively break the circuit when there is no speech on it. Noise contrast problems are caused by the background noise being interrupted when the circuit is broken so that the user listening on the circuit hears the background noise being continually switched on and off. This is particularly disturbing for a user talking to a user in a moving vehicle because the background noise being modulated in this way is at a very high level. In this situation, it has been found that speech intelligibility can be impaired. The main sources of background noise are: - background acoustic noise picked up by the microphone. For a loudspeaking telephone in a moving vehicle the speech/noise ratio can be as low as 0 db; - idle channel noise. This includes noise generated in the transmission system (thermal noise and crosstalk) the switching system and in speech transcoders.

23 Page Elements of a which can cause noise contrast impairment The following elements can cause noise contrast impairments: - the acoustic echo control device in the MS. A moving vehicle presents a very difficult environment for an echo canceller, so an echo suppressor is likely to be used (possibly in conjunction with an echo canceller). Echo suppressors contain voice-operated switches; - DTX. The transmitter switching will cause a PSTN user talking to a user to hear modulation of the mobile background noise. It will also cause the user to hear modulation of the PSTN noise. The PSTN noise will vary from connection to connection and should decrease in the future with increasing network digitalisation; - the electric echo control devices protecting the user against echo returned from the PSTN. The centre clipper in this echo canceller will cause some noise modulation Reduction of noise contrast A reduction in noise contrast: - reduces conversational difficulty, particularly for long conversations; - allows a greater tolerance on the matching of the level and spectrum of the comfort noise to the ambient noise. NOTE: Preliminary tests in vehicles indicate that, in a constant noise environment with a handsfree MS and a signal-to-noise ratio of approximately 10 db, a maximum level mismatch of 2 db can be tolerated. The comfort noise spectrum was a reconstruction of the averaged medium term ambient noise spectrum Reduction of noise contrast by limiting the noise received by the microphone The characteristics of the ambient noise (spectrum and level) depend on the environment in which the MS is used. As a microphone is characterised by its sensitivity and directivity, only part of this noise will enter the microphone. A general principle for reducing noise contrast is to maximise the signal-to-noise ratio at the microphone input. This can be achieved by simultaneously increasing directivity, reducing sensitivity, and placing the microphone close to the mouth of the talker. Consequently, the implementation of the acoustic terminal will significantly affect the dynamic range of the noise contrast Headset MS In the case of a headset and if DTX is disabled, then noise contrast will not be present since acoustic echo control (with centre clipping) is not required. If DTX is enabled, then only a small amount of noise contrast might result since the microphone would be close to the talker's mouth and would follow the movement of the talker's head, thus fulfilling the general principle described above. In the worst case, the headset is likely to give a minimum of 15 db signal-to-noise ratio. (This value is for further study) Handset MS In the case of a handset, and if DTX is disabled, then noise contrast will not be present if optimised echo cancelling techniques (without residual echo clipping) are used to control the acoustic echo (providing 46 db EL). If DTX is enabled or acoustic echo control with centre clipping is used, then only a small amount of noise contrast might result since the microphone would be close to the talker's mouth and would follow the movement of the talker's head, thus fulfilling the general principle described above. In the worst case, the handset is likely to give a minimum of 15 db signal-to-noise ratio. (This value is for further study).